Re: [OpenSIPS-Users] Opensips SIP trunk call handling
Hello, I sent you invitation on skype,nduwayezu Joselyne on skype, i'm waiting for your reply.I still have problems with the incoming call throught Opensips. I rely on your help please!! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602643.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
Thank you for your reply -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602582.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
You better ask this question on freeswitch list. Anyway you need to handle call in public.xml and add ip in acl too. On Apr 8, 2016 6:51 PM, "Francjos" <35...@heb.be> wrote: To tell Freeswitch that calls come from Opensips proxy, do i have to create a new external profile in sip_profiles directory or add an extension in dialplan/public.xml or both of two? Second question, in this file :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i have to specify the CIDR, is the ip address the one of Opensips? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602562.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
To tell Freeswitch that calls come from Opensips proxy, do i have to create a new external profile in sip_profiles directory or add an extension in dialplan/public.xml or both of two? Second question, in this file :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i have to specify the CIDR, is the ip address the one of Opensips? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602562.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
First, I'm gonna test the simple case without load balancing , i'll do changes after i success the simple routing. Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602560.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
No problem.
If that's all you want to do, then just include it within a route{} block
and you should be good to go.
I will say, of that's all you want to do, when you 'load balance the invite
to FS', FS will respond with a 100 trying - and at the minute your script
will reply to FS with a 405..
Perhaps you should spend some time reading about SIP, as it'll help you
more easily implement stuff on OpenSIPS
On Fri, Apr 8, 2016 at 10:12 AM, Francjos <35...@heb.be> wrote:
> Thank you for redirecting me to the tutorial. I picked up a part of what i
> need,i think i can adjust it to my needs but the problem i still have is
> where to include the logic of load balancing in opensips.cfg. The logic is
> the following:
>
> if (is_method("INVITE")) {
> if (!load_balance("1","pstn","1")) {
> send_reply("503","Service Unavailable");
> exit;
> }
> }
> else if (is_method("REGISTER")) {
> if (!ds_select_dst("1", "0")) {
> send_reply("503","Service Unavailable");
> exit;
> }
> }
> else {
> send_reply("405","Method Not Allowed");
> exit;
> }
>
> So that Opensips can routes calls to the write Freeswitch.
>
> Thanks again.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602558.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
Thank you for redirecting me to the tutorial. I picked up a part of what i
need,i think i can adjust it to my needs but the problem i still have is
where to include the logic of load balancing in opensips.cfg. The logic is
the following:
if (is_method("INVITE")) {
if (!load_balance("1","pstn","1")) {
send_reply("503","Service Unavailable");
exit;
}
}
else if (is_method("REGISTER")) {
if (!ds_select_dst("1", "0")) {
send_reply("503","Service Unavailable");
exit;
}
}
else {
send_reply("405","Method Not Allowed");
exit;
}
So that Opensips can routes calls to the write Freeswitch.
Thanks again.
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602558.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
Franjos, Have you read http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration ? Versions are a little out of date, but I think it looks correct still Ben On Fri, Apr 8, 2016 at 9:27 AM, Francjos <35...@heb.be> wrote: > Thanks .Can you please give me more details on how i have to proceed. > Thanks again > > > > -- > View this message in context: > http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602556.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
Thanks .Can you please give me more details on how i have to proceed. Thanks again -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552p7602556.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips SIP trunk call handling
You need to manage calls in freeswitch. On Apr 8, 2016 12:32 PM, "Francjos" <35...@heb.be> wrote: > Hello, > > I wish to configure Opensips that have to act as load balancer to > freeswitch > boxes. > When i directly connect freeswitch to the trunk, incoming calls are managed > in /usr/local/freeswitch/conf/dialplan/public/ directory. What about if i > use load balancing? Where do i have to manage incoming (and out ) calls? IN > Freeswitch or in Opensips? > > Thanks > > > > -- > View this message in context: > http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips SIP trunk call handling
Hello, I wish to configure Opensips that have to act as load balancer to freeswitch boxes. When i directly connect freeswitch to the trunk, incoming calls are managed in /usr/local/freeswitch/conf/dialplan/public/ directory. What about if i use load balancing? Where do i have to manage incoming (and out ) calls? IN Freeswitch or in Opensips? Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-SIP-trunk-call-handling-tp7602552.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
